[Asterisk-Users] dialplan defenition

[Joao Pereira]

But to have a transparent integration with VoIP and legacy, I cant make 
users dial twice... or having to whait for Asterisks dialtone, and dial 
the number.
I whant to dial the 74XXX from a PBX extension (74118 for example) and 
the IP phone rings.
Asterisk just need to forward the 74XXX calls, thats why I think the 
solution is close to this:

exten => _74XXX,1,Dial(SIP/${EXTEN}@193.136.252.5,30,r)

 ... but it always answers:
 pbx.c:1877 ast_pbx_run: Channel 'CAPI[contr1/118]/1' sent into invalid
 extension 's' in context 'default', but no invalid handler

Why is CAPI sending it to 's' if I explicitly write 
"Dial(SIP/${EXTEN}@193.136.252.5,30,r)" ??

João


Matt Riddell wrote:

> Joao Pereira wrote:
>
>> Hello list,
>> Im writing my dial plan, in witch every SIP phone begins with 74 and 
>> has more 3 numbers (like 74XXX).
>> So, I want to route all 74XXX calls to my sip channel. For this I 
>> wrote this line:
>> exten => s,1,Dial(SIP/74118 at 193.136.252.5,30,r)
>
>
> What is happening is that capi is sending it to s.
>
> You will need to either set up an IVR, asking which number to send it to.
>
> So, you would do the following:
>
> exten => s,1,Answer()
> exten => s,2,Background(pls-entr-extn)
> exten => _74XXX,1,Dial(SIP/${EXTEN})
> exten => _74XXX,2,Goto(s|1)
> exten => _74XXX,102,Goto(s|1)
>
> You will obviously need to record the pls-entr-extn sound.
>
> You can do this by making an exten like this:
>
> exten => 678,1,Record(pls-entr-extn)
>